WO2009076523A1 - Filtration adaptative dans un système à réseau de détecteurs - Google Patents

Filtration adaptative dans un système à réseau de détecteurs Download PDF

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Publication number
WO2009076523A1
WO2009076523A1 PCT/US2008/086407 US2008086407W WO2009076523A1 WO 2009076523 A1 WO2009076523 A1 WO 2009076523A1 US 2008086407 W US2008086407 W US 2008086407W WO 2009076523 A1 WO2009076523 A1 WO 2009076523A1
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WO
WIPO (PCT)
Prior art keywords
sensor array
filter
array device
microphones
input
Prior art date
Application number
PCT/US2008/086407
Other languages
English (en)
Inventor
Douglas Anderea
Leonard Shoell
Yehuda Mitelman
Original Assignee
Andrea Electronics Corporation
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Andrea Electronics Corporation filed Critical Andrea Electronics Corporation
Publication of WO2009076523A1 publication Critical patent/WO2009076523A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
    • G10L2021/02166—Microphone arrays; Beamforming

Definitions

  • microphones can be built into a computer or monitor, or may be an external device which is attached to a computer or monitor. Due to the distance between such microphones and the user, such microphones must be able to receive input from a greater area. As a consequence, such microphones are also subject to picking up increased background noise.
  • an object of the present invention provide for an integrated array of microphones utilizing an adaptive beam forming algorithm. Such an invention does not require an individual to wear a microphone headset and allows a large degree of freedom. Further, such a microphone array allows a user to electronically steer the microphone's beam, or the area in which it accepts voice input, as opposed to having to physically steer the microphone array.
  • the present invention relates to a sensor array having adaptive filtering capabilities and methods of using the same to reduce background and related noise.
  • the sensor array receives digital input from a number of channels. First an averaging filter is applied to the input of each channel. The signal-to-noise ratio (SNR) of the output of the averaging filter is calculated. Depending on the SNR, a second filter, namely an adaptive filter would then be applied to the output of the averaging filter. The coefficients of this adaptive filter are updated on the basis of several calculated parameters such as a calculation of the beam of the sensor, a beam reference, a reference average, and noise estimation. These calculations are done on a continuous basis and the adaptive filter coefficients are also continuously updated.
  • SNR signal-to-noise ratio
  • the averaging filter and adaptive filter may be implemented on a digital signal processor or DSP.
  • general microprocessors such as those found in computers maybe used to perform the digital processing to implement filtering.
  • the sensor array itself can be made of microphones. If analog microphones are used the input must be digitized before the digital filtering begins. Alternatively, Digital microelectromechanical systems (MEMS) microphones can be used, wherein the microphone itself digitizes the input.
  • MEMS microelectromechanical systems
  • the terms microphone array and sensor array are used interchangeably. Any embodiments described as referring to a microphone array are equally applicable to a sensor array, and vice versa.
  • Figure 1 is a drawing of a sensor array according to one embodiment of the invention.
  • Figure 2 is a schematic depicting the beam forming algorithm according to one embodiment of the invention.
  • Figure 3 is a drawing depicting a polar beam plot of a 2 member microphone array according to one embodiment of the invention.
  • Figure 4 is a drawing depicting the corresponding beam to the polar plot of Figure 3 according to one embodiment of the invention.
  • Figure 5 depicts a comparison between the filtering of Microsoft array filter with an array filter disclosed according to an embodiment of the present invention.
  • Figure 6 is a depiction of an example of a visual interface that can be used in accordance with the present invention.
  • Figure 7 is a depiction of an example of a visual interface that can be used in accordance with the present invention.
  • a sensor array receives signals from a source.
  • the digitized output of the sensors is then transformed using a discrete Fourier transform (DFT).
  • DFT discrete Fourier transform
  • the sensors of the sensor array preferably will consist of, but are not limited to, microphones.
  • the microphones will be aligned on a particular axis.
  • the array will comprise two microphones, 60 and 70 on a straight line axis.
  • the array will consist of an even amount of sensors, with the sensors, according to one embodiment, a fixed distance apart from each adjacent sensor.
  • the sensor array can be designed with a mount 80 to sit or attach to or on a computer monitor or similar.
  • a video camera 75 or some other type of device or sensor may fit or be located in-between the two most center microphones of the sensor array such that there is an equal amount of microphones on each side of the video camera or other device.
  • the microphones generally will be positioned horizontally, and symmetrically with respect to a vertical axis, hi such an arrangement there are two sets of microphones, one on each side of the vertical axis corresponding to two separate channels, a left and right channel, for example.
  • the microphones will be digital microphones such as uni or omni-directional electret microphones, or micro machined microelectromechanical systems (MEMS) microphones.
  • MEMS micro machined microelectromechanical systems
  • the MEMS microphones have silicon circuitry that internally converts an analog audio signal into a digital signal without the need of an A/D converter, as other microphones would require in other embodiments of this invention, hi any event, after the received audio signals are digitized, according to an embodiment of the present invention, the signals travel through adjustable delay lines that act as input into a microprocessor or a DSP.
  • the delay lines are adjustable, such that a user can control the beam of the array.
  • the delay lines are fed into the microprocessor of a computer.
  • GUI graphical user interface
  • the interface can tell the user the width of the beam produced from the array, the direction of the beam, and how much sound it is picking up from a source.
  • the user can vary the delay lines that carry the output of the digitizer or digital microphone to the microprocessor or DSP.
  • the delay lines As is well known in the areas of sensor array or antenna array technology, by changing the delay lines from the sensors, the direction of the beam can be changed. This allows a user then to steer the beam.
  • the microphone array might by default produce a beam direction that is directly straightforward from the microphone array.
  • the target signal is not directly ahead of the sensor array, but instead at an angle with respect to the sensor array, it would extremely helpful for the user to steer the beam in the direction of the target source. Allowing a person to steer the beam through electronic beams is more efficient than requiring the manual movement of the device containing the sensor array.
  • the steering ability allows the sensor array, including a microphone array, itself to be small and compact without requiring parts to physically move the sensors.
  • the software receiving the input would process the input through the GUI and properly translate the commands of user to accordingly adjust the delay lines to the user's wishes.
  • the beam may be steered before any input or anytime after the sensor array or microphones receive input from a source.
  • the present invention produces substantial cancellation or reduction of background noise.
  • the steerable microphone array produces a two-channel input signal that is digitized 20 and on which beam steering is applied 22, the output is transformed using a DFT 24.
  • a DFT fast Fourier transform
  • FFT fast Fourier transform
  • the DFT processing can take place in a general microprocessor, or a DSP.
  • the data can be filtered according to the embodiment of Figure 2. This invention applies an adaptive filter in order to greatly filter out background noise.
  • the key is the way in which the adaptive filter is composed and in particular how the coefficients that make up the filter are produced.
  • the adaptive filter is a mathematical transfer function.
  • the filter coefficient is dependent on the past and present digital input.
  • An embodiment as shown in Figure 2 discloses an averaging filter that is first applied to the digitally transformed input in order to smooth the digital input and remove high frequency artifacts 26. This is done for each channel.
  • the noise from each channel is also determined 28. Once the noise is determined, different variables can be calculated to update the adaptive filter coefficients.
  • the channels are averaged and compared against a calibration threshold 32. Such a threshold is usually set by the manufacturer. If the result falls below a threshold, the values are adjusted by a weighting average function such as to reduce distortion by a phase mismatch between the channels.
  • SNR signal to noise ratio
  • the SNR is calculated from the averaging filter output and the noise calculated 34 from each channel.
  • the result of the SNR calculation if it reaches a certain threshold will trigger modifying the digital input using the filter coefficients of the previous calculated beam.
  • the threshold which is typically set by the manufacturer, is a value in which the output may be sufficiently reliable for use in certain applications. In different situations or applications, a higher SNR may be desired, and the threshold may be adjusted by an individual.
  • the beam for each input is continuously calculated.
  • a beam is calculated as the average of signals, for instance, of two signals from a left and right channel, the average including the difference of angle between the target source and each channel.
  • a beam reference, reference average, and beam average are also calculated 36.
  • the beam reference is a weighted average of a previous calculated beam and the adaptive filter coefficients.
  • a reference average is the weighted sum of the previous calculated beam references.
  • beam average which is the running average of previous calculated beams. All these factors are used to update the adaptive filter.
  • an error calculation is performed by subtracting the current beam from the beam average 42. This error is then used in conjunction with an updated reference average 44 and updated beam average 40 in a noise estimation calculation 46.
  • the noise calculation helps predict the noise from the system including the filter.
  • the noise prediction calculation is used in updating the coefficients of the adaptive filter 48 such as to minimize or eliminate potential noise.
  • the output of the filter is then processed by an inverse discrete Fourier transform (IDFT).
  • IDFT inverse discrete Fourier transform
  • the output then may be used in digital form as input into an audio application, such as audio recording, voice over internet protocol (VOIP), speech recognition, or the output can be sent as input to another, separate computing system for additional processing.
  • an audio application such as audio recording, voice over internet protocol (VOIP), speech recognition, or the output can be sent as input to another, separate computing system for additional processing.
  • VOIP voice over internet protocol
  • speech recognition or the output can be sent as input to another, separate computing system for additional processing.
  • the digital output from the adaptive filter may be reconverted by a D/A converter into an analog signal and sent to an output device.
  • the output from the filter can be sent as input to another computer or electronic device for processing. Or it may be sent to an acoustic device such as a speaker system, or headphones for example.
  • the algorithm is advantageously able to effectively filtering of noise, including non-stationary noise or sudden noise such as a door slamming. Furthermore, the algorithm allows superior filtering at lower frequencies while also allowing the spacing between elements in the array, i.e., between microphones, to be small, including as little as 2 inches or 50 mm in a two element microphone embodiment. Previously, microphones arrays would require substantially greater spacing, such as a foot or more between elements to be able to have the same amount filtering at the lower frequencies. Another advantage of the algorithm as presented is that it, for the most part, requires no customization for a wide range of different spacings between the elements in the array. The algorithm is robust and flexible enough to automatically adjust and handle the element spacing a microphone array system might be required to have in order to work in conjunction with common electronic or computer devices.
  • Figure 3 shows a polar beam plot of a 2 member microphone array according to an embodiment of the invention wherein the delays lines of the left and right channels are equal.
  • Figure 4 shows the corresponding beam as shown in the polar plot of Figure 3 in an embodiment where the microphone array is used in conjunction with a computer system.
  • the microphone array is placed a top a monitor in Figure 4.
  • the speakers are placed outside of the main beam. Because of the superior performance of the microphone array system, the array attenuates signals originating from sources outside of the main beam, such as the speakers as shown in Figure 4, such that microphone array effectively acts as an echo canceller with there being no feedback distortion.
  • the beam typically will be focused narrowly on the target source, which is typically the human voice, as depicted in Figure 4.
  • the input of the microphone array shows a dramatic decrease in signal strength as shown in Figure 5.
  • the 12,000 mark on the axis represents a target source or input source directly in front of the microphone array.
  • the 10,000 mark and 14,000 mark correspond to the outer parts of the beam as shown in Figure 3 and 3.
  • Figure 5 shows, for example, a comparison between the filtering of a Microsoft array filter with an array filter according to an embodiment of the present invention.
  • the target source falls outside of the beam width, or at the 10,000 or 14,000 marks, there is a very noticeable and dramatic roll off in signal strength in the microphone array using an embodiment of the present invention. By contrast, there is no such roll off found in the Microsoft array filter.
  • the sensor array could be placed on or integrated within different types of devices such as any devices that require or may use an audio input, such a computer system, laptop, cellphone, global positioning system, audio recorder, etc.
  • the microphone array may be integrated, wherein the signals from the microphones are carried through delay lines directly into the computer's microprocessor.
  • the calculations performed for the algorithm described according to an embodiment of the present invention may take place in a microprocessor, such as an Intel Pentium Processor, typically used for personal computers.
  • the processing may be done by a digital signal processor (DSP).
  • DSP digital signal processor
  • the microprocessor or DSP may be used to handle the user input to control the adjustable lines and the beam steering.
  • the microphone array and the delay lines can be connected, for example, to a USB input instead of being integrated with a computer system.
  • the signals may then be routed to the microprocessor, or it may be routed to a separate DSP chip that is also connected to the same or different computer system for digital processing.
  • the microprocessor of the computer in such an embodiment could still run the GUI that allows the user to control the delays and thus control the steering of the beam, but the DSP will perform the appropriate filtering of the signal according to an embodiment of an algorithm presented herein.
  • the spacing of the microphones in the sensor array maybe adjustable. By adjusting the spacing, the directivity and beam width of the sensor can be modified. In some embodiments, if a video sensor or camera is placed in the center of the microphone array it may be preferable to have the beam width the same as the optical viewing angle of the video camera or sensor.
  • Figures 6 and-7 depict alternate visual user interfaces that be used with the invention as disclosed.
  • Figure 7 is a portion of the visual interface as shown in Figure 5.

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Obtaining Desirable Characteristics In Audible-Bandwidth Transducers (AREA)

Abstract

La présente invention concerne un réseau de détecteurs orientable qui reçoit une entrée d'une cible et applique un filtre de moyenne. Un filtre adaptatif est alors utilisé si le SNR de la sortie du filtre de moyenne atteint un seuil.
PCT/US2008/086407 2007-12-11 2008-12-11 Filtration adaptative dans un système à réseau de détecteurs WO2009076523A1 (fr)

Applications Claiming Priority (2)

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US1288407P 2007-12-11 2007-12-11
US61/012,884 2007-12-11

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